Cisco SPA8800 Configuring page 55

Ip telephony gateway
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Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment
Supported: replaces
Contact: <sip:85551313@192.168.2.20>
Contact Binding: <sip:85551313@192.168.2.20>
Content-Length: 0
Frame 6 (1036 bytes on wire, 1036 bytes captured)
Ethernet II, Src: Internet_1c:33:a1 (00:e0:4d:1c:33:a1), Dst: 00:24:97:f0:50:3c
(00:24:97:f0:50:3c)
Internet Protocol, Src: 192.168.2.20 (192.168.2.20), Dst: 192.168.2.237 (192.168.2.237)
User Datagram Protocol, Src Port: sip (5060), Dst Port: 5161 (5161)
Session Initiation Protocol
Request-Line: INVITE sip:5551313@192.168.2.237:5161 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK18bc3f3d;rport
Transport: UDP
Sent-by Address: 192.168.2.20
Sent-by port: 5060
Branch: z9hG4bK18bc3f3d
RPort: rport
From: "SPA8k8Phone1" <sip:101@192.168.2.20>;tag=as7b52add2
SIP Display info: "SPA8k8Phone1"
SIP from address: sip:101@192.168.2.20
SIP tag: as7b52add2
To: <sip:5551313@192.168.2.237:5161>
SIP to address: sip:5551313@192.168.2.237:5161
Contact: <sip:101@192.168.2.20>
Contact Binding: <sip:101@192.168.2.20>
Call-ID: 3db967441067a5b61785c0606e95f7a0@192.168.2.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 05 Jun 2009 20:26:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 453
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 7519 7519 IN IP4 192.168.2.20
Session Name (s): session
Connection Information (c): IN IP4 192.168.2.20
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 13440 RTP/AVP 0 3 8 112 5 10 7 110
111 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:112 AAL2-G726-32/8000
Media Attribute (a): rtpmap:5 DVI4/8000
Media Attribute (a): rtpmap:10 L16/8000
Media Attribute (a): rtpmap:7 LPC/8000
Media Attribute (a): rtpmap:110 speex/8000
Media Attribute (a): rtpmap:111 G726-32/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv
Frame 7 (497 bytes on wire, 497 bytes captured)
Ethernet II, Src: Internet_1c:33:a1 (00:e0:4d:1c:33:a1), Dst: 00:24:97:f0:50:3c
(00:24:97:f0:50:3c)
Internet Protocol, Src: 192.168.2.20 (192.168.2.20), Dst: 192.168.2.237 (192.168.2.237)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 180 Ringing
Message Header
Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK-2bd81ced;received=192.168.2.237
Transport: UDP
Sent-by Address: 192.168.2.237
Sent-by port: 5060
Branch: z9hG4bK-2bd81ced
Received: 192.168.2.237
From: "SPA8k8Phone1" <sip:101@192.168.2.20>;tag=432331171f77868do0
SIP Display info: "SPA8k8Phone1"
SIP from address: sip:101@192.168.2.20
SIP tag: 432331171f77868do0
To: <sip:85551313@192.168.2.20>;tag=as62288bf1
SIP to address: sip:85551313@192.168.2.20
SIP tag: as62288bf1
Call-ID: c6e3a35f-e0a478e5@192.168.2.237
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
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